Telephony over packet networks, particularly over wide area networks such as the Internet, has received considerable attention recently. Specifically, it is now possible to route voice, data, and video over data networks such as the Internet, and long distance telephone calls may now be routinely routed over the Internet. The transmission of such information streams over packet switched networks such as the Internet can be more cost efficient than traditional telephony, which requires a dedicated circuit between the calling and called party.
Computer telephone integration (CTI) has been widely applied to traditional telephony methods and apparatus, but has not as yet been successfully applied to the methods and apparatus used for packet network telephony.
More specifically, although it is common to control a Private Branch Exchange (PBX) or similar equipment from a remote applications computer, the use of such remote applications computers is in its early stages in the field of packet telephony.
Several protocols exist that define methods and apparatus to convey calls over packet switched data networks, such as the Internet. One possible set of methods and apparatus for completing calls over a packet switched network, which calls can handle voice and other information streams, is defined by ITU recommendation H.323. The H.323 standard is available from the International Telecommunications Unit, Geneva and is well known in the relevant industry. The H.323 standard defines various protocols dealing with call control, call setup, call termination, and other similar techniques known to those in the packet network telephony art.
The H.323 standard defines a functional entity called a gatekeeper. The gatekeeper handles network functions such as bandwidth control, zone management, address translation, and admissions control for a designated set of network terminals.
The gatekeeper function provides services analogous to the call processing function within a private branch exchange (PBX) in conventional telephony. In traditional telephony, CTI features are provided by creating an interface between external application software and the call processing function within the PBX. By contrast, in the known packet network telephony art, interfaces to the gatekeeper are only defined from other gatekeepers, end points, and other network entities.
A second protocol for performing packet telephony control functions is the Session Initiation Protocol (SIP) defined by IETF RFC 2543. Like H.323, SIP includes a plurality of functions that establish, modify and terminate multimedia sessions. The SIP methodologies include a variety of functions that go beyond those described by H.323. For example, SIP includes the ability to support mobile users, the use of standardized HTTP syntax and URLs, the ability to have multiple SIP connections through a single TCP/IP session, the use of “proxy servers” (defined further below) and a variety of other robust features discussed in more detail below.
However, SIP methodologies also lack the ability to seamlessly interface to a third party applications computer for the purpose of implementing call control, monitoring, and related functions. Additionally, SIP is utilizes a connectionless protocol, making call setup much faster.